<div dir="ltr">Hi Everyone,<br>Thanks for your reply,<br>I have added 'tT' options in my dialplan and also agi script which is handles the outbound and inbound calls, <br>i have also set the transfer context like "__TRANSFER_CONTEXT=outgoing".<br>
Still inbound call has transfer issue!<br><br>Please help me for this issue.<br><br><br>-- <br>Thanks,<br>Max Alex<br>Voip Developer<br><br><div class="gmail_quote">On Mon, Aug 25, 2008 at 7:30 PM, Serge Berney <span dir="ltr"><<a href="mailto:s.berney@kinonline.net">s.berney@kinonline.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div link="blue" vlink="purple" lang="FR-CH">
<div>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">I experiment the same problem… On some call, there's
no possibility to transfer the call (caller ear DTMF sounds) – AST <a href="http://1.4.21." target="_blank">1.4.21.</a></span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">For Max : </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">DialPlan must be like this :</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p style="text-indent: 35.4pt;"><span style="font-size: 11pt; color: rgb(31, 73, 125);">exten => 99, 1, Answer()</span></p>
<p style="text-indent: 35.4pt;"><span style="font-size: 11pt; color: rgb(31, 73, 125);">exten => 99, n, Dial(SIP/sip_user1,
20, tT)</span></p>
<p style="text-indent: 35.4pt;"><span style="font-size: 11pt; color: rgb(31, 73, 125);">exten => 99, 1, Hangup()</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Regards</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<div>
<div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0cm 0cm;">
<p><b><span style="font-size: 10pt;" lang="FR">De :</span></b><span style="font-size: 10pt;" lang="FR">
<a href="mailto:asterisk-dev-bounces@lists.digium.com" target="_blank">asterisk-dev-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-dev-bounces@lists.digium.com" target="_blank">asterisk-dev-bounces@lists.digium.com</a>]
<b>De la part de</b> Fernando Urzedo<br>
<b>Envoyé :</b> lundi, 25. août 2008 13:49<br>
<b>À :</b> Asterisk Developers Mailing List<br>
<b>Objet :</b> Re: [asterisk-dev] Blind Transfer is not working in
incoming calls</span></p>
</div>
</div><div><div></div><div class="Wj3C7c">
<p> </p>
<p><span style="font-size: 10pt; color: blue;">Hi Max,</span></p>
<p> </p>
<p><span style="font-size: 10pt; color: blue;">Please make sure you are setting the Dial command with parameters
"T" and "t". I would say you are missing "t"...</span></p>
<p> </p>
<p><span style="font-size: 10pt; color: blue;">Regards!</span></p>
<p> </p>
<div style="text-align: center;" align="center"><span lang="EN-US">
<hr align="center" size="2" width="100%">
</span></div>
<p style="margin-bottom: 12pt;"><b><span style="font-size: 10pt;" lang="EN-US">From:</span></b><span style="font-size: 10pt;" lang="EN-US">
<a href="mailto:asterisk-dev-bounces@lists.digium.com" target="_blank">asterisk-dev-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-dev-bounces@lists.digium.com" target="_blank">asterisk-dev-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Max Alex<br>
<b>Sent:</b> sábado, 23 de agosto de 2008 03:53<br>
<b>To:</b> <a href="mailto:asterisk-dev@lists.digium.com" target="_blank">asterisk-dev@lists.digium.com</a><br>
<b>Subject:</b> [asterisk-dev] Blind Transfer is not working in incoming calls</span><span lang="EN-US"></span></p>
<div>
<p style="margin-bottom: 12pt;">Hi Everybody,<br>
i have installed asterisk 1.4.19 on my box,<br>
I have setup agi script which is used while incoming and outgoing calls.<br>
It will find the users for incoming and calls to them which is registered in
asterisk,<br>
I have a setup *# for blind transfer to call any outbound or inbound numbers.<br>
when i am calling any outbound call and the calls are connected with my sip
peer, then i am pressing *# for blind transfer, it will ask me to enter the
transfer number and it is working,<br>
But when an incoming call to my sip user and they are connected the *# is not
worked even the transfer prompt is also played, and dtmf is also set properly.<br>
<br>
But i am not getting why the incoming call is not transfer to any other number?<br>
Please help for this issue!<br clear="all">
<br>
<br>
-- <br>
Thanks,<br>
Max Alex<br>
Voip Developer</p>
</div>
</div></div></div>
</div>
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